Chapter 7 Multimedia Networking

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7: Multimedia Networking 7-1

Chapter 7 Multimedia Networking

A note on the use of these ppt slides:

We’re making these slides freely available to all (faculty, students, readers).

They’re in PowerPoint form so you can add, modify, and delete slides (including this one) and slide content to suit your needs. They obviously represent a lot of work on our part. In return for use, we only ask the following:

 If you use these slides (e.g., in a class) in substantially unaltered form, that you mention their source (after all, we’d like people to use our book!)

 If you post any slides in substantially unaltered form on a www site, that you note that they are adapted from (or perhaps identical to) our slides, and note our copyright of this material.

Thanks and enjoy! JFK / KWR

Computer Networking: A Top Down Approach

5^th edition.

Jim Kurose, Keith Ross Addison-Wesley, April 2009.

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Multimedia and Quality of Service: What is it?

multimedia applications:

network audio and video (“continuous media”)

network provides

application with level of performance needed for application to function.

QoS

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Chapter 7: goals

Principles

 classify multimedia applications

 identify network services applications need

 making the best of best effort service Protocols and Architectures

 specific protocols for best-effort

 mechanisms for providing QoS

 architectures for QoS

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Chapter 7 outline

7.1 multimedia networking applications

7.2 streaming stored audio and video

7.3 making the best out of best effort service

7.4 protocols for real-time interactive applications

RTP,RTCP,SIP

7.5 providing multiple classes of service 7.6 providing QoS

guarantees

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MM Networking Applications

Fundamental

characteristics:

 typically delay sensitive

 end-to-end delay

 delay jitter

 loss tolerant: infrequent losses cause minor

glitches

 antithesis of data, which are loss intolerant but delay tolerant.

Classes of MM applications:

1) stored streaming 2) live streaming

3) interactive, real-time

Jitter is the variability of packet delays within the same packet stream

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Streaming Stored Multimedia

Stored streaming:

 media stored at source

 transmitted to client

 streaming: client playout begins before all data has arrived

 timing constraint for still-to-be

transmitted data: in time for playout

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Streaming Stored Multimedia:

What is it?

1. video recorded

2. video

sent 3. video received,

played out at client

streaming: at this time, client playing out early part of video, while server still sending later part of video

network delay

time

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Streaming Stored Multimedia: Interactivity

 VCR-like functionality: client can pause, rewind, FF, push slider bar

 10 sec initial delay OK

 1-2 sec until command effect OK

 timing constraint for still-to-be

transmitted data: in time for playout

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Streaming Live Multimedia

Examples:

 Internet radio talk show

 live sporting event

Streaming (as with streaming stored multimedia)

 playback buffer

 playback can lag tens of seconds after transmission

 still have timing constraint Interactivity

 fast forward impossible

 rewind, pause possible!

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Real-Time Interactive Multimedia

 end-end delay requirements:

 audio: < 150 msec good, < 400 msec OK

• includes application-level (packetization) and network delays

• higher delays noticeable, impair interactivity

 session initialization

 how does callee advertise its IP address, port number, encoding algorithms?

 applications: IP telephony,

video conference, distributed interactive worlds

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Multimedia Over Today’s Internet

TCP/UDP/IP: “best-effort service”

 no guarantees on delay, loss

Today’s Internet multimedia applications use application-level techniques to mitigate

(as best possible) effects of delay, loss But you said multimedia apps requires

QoS and level of performance to be effective!

? ? ?

? ?

?

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How should the Internet evolve to better support multimedia?

Integrated services philosophy:

 fundamental changes in Internet so that apps can reserve end-to-end

bandwidth

 requires new, complex

software in hosts & routers Laissez-faire

 no major changes

 more bandwidth when needed

 content distribution,

application-layer multicast

 application layer

Differentiated services philosophy:

 fewer changes to Internet infrastructure, yet provide 1st and 2nd class service

What’s your opinion?

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A few words about audio compression

 analog signal sampled at constant rate

 telephone: 8,000 samples/sec

 CD music: 44,100 samples/sec

 each sample quantized, i.e., rounded

 e.g., 2⁸=256 possible quantized values

 each quantized value represented by bits

 8 bits for 256 values

 example: 8,000 samples/sec, 256 quantized values -->

64,000 bps

 receiver converts bits back to analog signal:

 some quality reduction

Example rates

 CD: 1.411 Mbps

 MP3: 96, 128, 160 kbps

 Internet telephony:

5.3 kbps and up

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A few words about video compression

 video: sequence of images displayed at constant rate

 e.g. 24 images/sec

 digital image: array of pixels

 each pixel represented by bits

 redundancy

 spatial (within image)

 temporal (from one image to next)

Examples:

 MPEG 1 (CD-ROM) 1.5 Mbps

 MPEG2 (DVD) 3-6 Mbps

 MPEG4 (often used in Internet, < 1 Mbps) Research:

 layered (scalable) video

 adapt layers to available bandwidth

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Chapter 7 outline

RTP,RTCP,SIP

guarantees

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Streaming Stored Multimedia

application-level streaming techniques for making the best out of best effort service:

 client-side buffering

 use of UDP versus TCP

 multiple encodings of multimedia

 jitter removal

 decompression

 error concealment

 graphical user interface w/ controls for

interactivity

Media Player

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Internet multimedia: simplest approach

audio, video not streamed:

 no, “pipelining,” long delays until playout!

 audio or video stored in file

 files transferred as HTTP object

 received in entirety at client

 then passed to player

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Internet multimedia: streaming approach

 browser GETs metafile

 browser launches player, passing metafile

 player contacts server

 server streams audio/video to player

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Streaming from a streaming server

 allows for non-HTTP protocol between server, media player

 UDP or TCP for step (3), more shortly

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constant bit rate video transmission

time variable

network delay

client video

reception constant bit rate video playout at client

client playout delay

buffered video

Streaming Multimedia: Client Buffering

 client-side buffering, playout delay compensate for network-added delay, delay jitter

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Streaming Multimedia: Client Buffering

 client-side buffering, playout delay compensate for network-added delay, delay jitter

buffered video variable fill

rate, x(t)

constant drain rate, d

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Streaming Multimedia: UDP or TCP?

UDP

 server sends at rate appropriate for client (oblivious to network congestion !)

 often send rate = encoding rate = constant rate

 then, fill rate = constant rate - packet loss

 short playout delay (2-5 seconds) to remove network jitter

 error recover: time permitting

TCP

 send at maximum possible rate under TCP

 fill rate fluctuates due to TCP congestion control

 larger playout delay: smooth TCP delivery rate

 HTTP/TCP passes more easily through firewalls

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Streaming Multimedia: client rate(s)

Q: how to handle different client receive rate capabilities?

 28.8 Kbps dialup

 100 Mbps Ethernet

A: server stores, transmits multiple copies of video, encoded at different rates

1.5 Mbps encoding 28.8 Kbps encoding

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User Control of Streaming Media: RTSP

HTTP

 does not target

multimedia content

 no commands for fast forward, etc.

RTSP: RFC 2326

 client-server

application layer protocol

 user control: rewind, fast forward, pause, resume, repositioning, etc…

What it doesn’t do:

 doesn’t define how audio/video is

encapsulated for

streaming over network

 doesn’t restrict how streamed media is transported (UDP or TCP possible)

 doesn’t specify how media player buffers audio/video

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RTSP: out of band control

FTP uses an “out-of-

band” control channel:

 file transferred over one TCP connection.

 control info (directory changes, file deletion, rename) sent over

separate TCP connection

 “out-of-band”, “in- band” channels use different port

numbers

RTSP messages also sent out-of-band:

 RTSP control messages use different port

numbers than media stream: out-of-band.

 port 554

 media stream is

considered “in-band”.

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RTSP Example

Scenario:

 metafile communicated to web browser

 browser launches player

 player sets up an RTSP control connection, data connection to streaming server

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Metafile Example

<title>Twister</title>

<track type=audio

e="PCMU/8000/1"

src = "rtsp://audio.example.com/twister/audio.en/lofi">

<track type=audio

e="DVI4/16000/2" pt="90 DVI4/8000/1"

src="rtsp://audio.example.com/twister/audio.en/hifi">

</switch>

<track type="video/jpeg"

src="rtsp://video.example.com/twister/video">

</group>

</session>

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RTSP Operation

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RTSP Exchange Example

C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Transport: rtp/udp; compression; port=3056; mode=PLAY S: RTSP/1.0 200 1 OK

Session 4231

C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231

Range: npt=0-

C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231

Range: npt=37

C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231

S: 200 3 OK

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Chapter 7 outline

RTP,RTCP,SIP

guarantees

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Real-time interactive applications

 PC-2-PC phone

 Skype

 PC-2-phone

 Dialpad

 Net2phone

 Skype

 videoconference with webcams

 Skype

 Polycom

Going to now look at a PC-2-PC Internet phone example in detail

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Interactive Multimedia: Internet Phone

Introduce Internet Phone by way of an example

 speaker’s audio: alternating talk spurts, silent periods.

 64 kbps during talk spurt

 pkts generated only during talk spurts

 20 msec chunks at 8 Kbytes/sec: 160 bytes data

 application-layer header added to each chunk.

 chunk+header encapsulated into UDP segment.

 application sends UDP segment into socket every 20 msec during talkspurt

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Internet Phone: Packet Loss and Delay

 network loss: IP datagram lost due to network congestion (router buffer overflow)

 delay loss: IP datagram arrives too late for playout at receiver

 delays: processing, queueing in network; end- system (sender, receiver) delays

 typical maximum tolerable delay: 400 ms

 loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10%

can be tolerated.

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constant bit rate transmission

time variable

network delay (jitter)

client

reception constant bit rate playout at client

client playout delay

buffered data

Delay Jitter

 consider end-to-end delays of two consecutive packets: difference can be more or less than 20 msec (transmission time difference)

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Internet Phone: Fixed Playout Delay

 receiver attempts to playout each chunk exactly q msecs after chunk was generated.

 chunk has time stamp t: play out chunk at t+q .

 chunk arrives after t+q: data arrives too late for playout, data “lost”

 tradeoff in choosing q:

 large q: less packet loss

 small q: better interactive experience

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Fixed Playout Delay

packets

time

packets generated

packets received

loss

r

p p'

playout schedule p' - r

playout schedule p - r

• sender generates packets every 20 msec during talk spurt.

• first packet received at time r

• first playout schedule: begins at p

• second playout schedule: begins at p’

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Adaptive Playout Delay (1)

packet ith

receiving after

delay network

average of

estimate d

acket p

ith for delay network

t r

receiver at

played is

i packet time

the p

receiver by

received is

i packet time

the r

packet ith

the of timestamp t

i i i

i i

i







dynamic estimate of average delay at receiver:

) (

) 1

( _i ₁ _i _i

i u d u r t

d   _  

where u is a fixed constant (e.g., u = .01).

 Goal: minimize playout delay, keeping late loss rate low

 Approach: adaptive playout delay adjustment:

 estimate network delay, adjust playout delay at beginning of each talk spurt.

 silent periods compressed and elongated.

 chunks still played out every 20 msec during talk spurt.

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Adaptive playout delay (2)

 also useful to estimate average deviation of delay, v_i:

|

| )

1

( _i ₁ _i _i _i

i u v u r t d

v   _   

 estimates d_i , v_i calculated for every received packet (but used only at start of talk spurt

 for first packet in talk spurt, playout time is:

i i

i

i t d Kv

p   

where K is positive constant

 remaining packets in talkspurt are played out periodically

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Adaptive Playout (3)

Q: How does receiver determine whether packet is first in a talkspurt?

 if no loss, receiver looks at successive timestamps.

 difference of successive stamps > 20 msec -->talk spurt begins.

 with loss possible, receiver must look at both time stamps and sequence numbers.

 difference of successive stamps > 20 msec and sequence numbers without gaps --> talk spurt begins.

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Recovery from packet loss (1)

Forward Error Correction (FEC): simple scheme

 for every group of n

chunks create redundant chunk by exclusive OR-ing n original chunks

 send out n+1 chunks,

increasing bandwidth by factor 1/n.

 can reconstruct original n chunks if at most one lost chunk from n+1 chunks

 playout delay: enough time to receive all n+1 packets

 tradeoff:

 increase n, less bandwidth waste

 increase n, longer playout delay

 increase n, higher probability that 2 or more chunks will be lost

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Recovery from packet loss (2)

2nd FEC scheme

 “piggyback lower quality stream”

 send lower resolution audio stream as

redundant information

 e.g., nominal

stream PCM at 64 kbps and redundant stream GSM at 13 kbps.

 whenever there is non-consecutive loss, receiver can conceal the loss.

 can also append (n-1)st and (n-2)nd low-bit rate chunk

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Recovery from packet loss (3)

Interleaving

 chunks divided into smaller units

 for example, four 5 msec units per chunk

 packet contains small units from different chunks

 if packet lost, still have most of every chunk

 no redundancy overhead, but increases playout delay

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Content distribution networks (CDNs)

Content replication

 challenging to stream large files (e.g., video) from single origin server in real time

 solution: replicate content at hundreds of servers

throughout Internet

 content downloaded to CDN servers ahead of time

 placing content “close” to user avoids impairments (loss, delay) of sending content over long paths

 CDN server typically in edge/access network

origin server in North America

CDN distribution node

CDN server

in S. America CDN server in Europe

CDN server in Asia

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Content distribution networks (CDNs)

Content replication

 CDN (e.g., Akamai)

customer is the content provider (e.g., CNN)

 CDN replicates

customers’ content in CDN servers.

 when provider updates content, CDN updates servers

origin server in North America

CDN distribution node

CDN server

in S. America CDN server in Europe

CDN server in Asia

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CDN example

origin server (www.foo.com)

 distributes HTML

 replaces:

http://www.foo.com/sports.ruth.gif

with

http://www.cdn.com/www.foo.com/sports/ruth.gif

HTTP request for

www.foo.com/sports/sports.html

DNS query for www.cdn.com

HTTP request for

www.cdn.com/www.foo.com/sports/ruth.gif 1

2 3

origin server

CDN’s authoritative DNS server

CDN server near client

CDN company (cdn.com)

 distributes gif files

 uses its authoritative DNS server to route redirect requests

client

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More about CDNs

routing requests

 CDN creates a “map”, indicating distances from leaf ISPs and CDN nodes

 when query arrives at authoritative DNS server:

 server determines ISP from which query originates

 uses “map” to determine best CDN server

 CDN nodes create application-layer overlay network

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Summary: Internet Multimedia: bag of tricks

 use UDP to avoid TCP congestion control (delays) for time-sensitive traffic

 client-side adaptive playout delay: to compensate for delay

 server side matches stream bandwidth to available client-to-server path bandwidth

 chose among pre-encoded stream rates

 dynamic server encoding rate

 error recovery (on top of UDP)

 FEC, interleaving, error concealment

 retransmissions, time permitting

 CDN: bring content closer to clients

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Chapter 7 outline

RTP, RTCP, SIP

guarantees

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Real-Time Protocol (RTP)

 RTP specifies packet structure for packets carrying audio, video data

 RFC 3550

 RTP packet provides

 payload type identification

 packet sequence numbering

 time stamping

 RTP runs in end systems

 RTP packets

encapsulated in UDP segments

 interoperability: if two Internet phone

applications run RTP, then they may be able to work together

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RTP runs on top of UDP

RTP libraries provide transport-layer interface that extends UDP:

• port numbers, IP addresses

• payload type identification

• packet sequence numbering

• time-stamping

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RTP Example

 consider sending 64 kbps PCM-encoded voice over RTP.

 application collects encoded data in

chunks, e.g., every 20 msec = 160 bytes in a chunk.

 audio chunk + RTP header form RTP packet, which is

encapsulated in UDP segment

 RTP header indicates type of audio encoding in each packet

 sender can change encoding during conference.

 RTP header also contains sequence

numbers, timestamps.

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RTP and QoS

 RTP does not provide any mechanism to ensure timely data delivery or other QoS guarantees.

 RTP encapsulation is only seen at end systems (not) by intermediate routers.

 routers providing best-effort service, making no special effort to ensure that RTP packets arrive at destination in timely matter.

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RTP Header

Payload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in middle of conference, sender

informs receiver via payload type field.

•Payload type 0: PCM mu-law, 64 kbps

•Payload type 3, GSM, 13 kbps

•Payload type 7, LPC, 2.4 kbps

•Payload type 26, Motion JPEG

•Payload type 31. H.261

•Payload type 33, MPEG2 video

Sequence Number (16 bits): Increments by one for each RTP packet sent, and may be used to detect packet loss and to restore packet

sequence.

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RTP Header (2)

 Timestamp field (32 bytes long): sampling instant of first byte in this RTP data packet

 for audio, timestamp clock typically increments by one for each sampling period (for example, each 125 usecs for 8 KHz sampling clock)

 if application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive.

 SSRC field (32 bits long): identifies source of t RTP stream. Each stream in RTP session should have distinct SSRC.

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RTSP/RTP Programming Assignment

 build a server that encapsulates stored video frames into RTP packets

 grab video frame, add RTP headers, create UDP segments, send segments to UDP socket

 include seq numbers and time stamps

 client RTP provided for you

 also write client side of RTSP

 issue play/pause commands

 server RTSP provided for you

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Real-Time Control Protocol (RTCP)

 works in conjunction with RTP.

 each participant in RTP session periodically

transmits RTCP control packets to all other

participants.

 each RTCP packet

contains sender and/or receiver reports

 report statistics useful to application: # packets

sent, # packets lost, interarrival jitter, etc.

 feedback can be used to control

performance

 sender may modify its transmissions based on feedback

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RTCP - Continued

 each RTP session: typically a single multicast address; all RTP /RTCP packets belonging to session use multicast address.

 RTP, RTCP packets distinguished from each other via distinct port numbers.

 to limit traffic, each participant reduces RTCP traffic as number of conference participants increases

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RTCP Packets

Receiver report packets:

 fraction of packets lost, last sequence number, average interarrival jitter

Sender report packets:

 SSRC of RTP stream, current time, number of packets sent, number of bytes sent

Source description packets:

 e-mail address of

sender, sender's name, SSRC of associated RTP stream

 provide mapping

between the SSRC and the user/host name

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Synchronization of Streams

 RTCP can synchronize different media streams within a RTP session

 consider videoconferencing app for which each sender generates one RTP stream for video, one for audio.

 timestamps in RTP packets tied to the video, audio sampling clocks

 not tied to wall-clock time

 each RTCP sender-report packet contains (for most recently generated packet in associated RTP stream):

 timestamp of RTP packet

 wall-clock time for when packet was created.

 receivers uses association to synchronize playout of audio, video

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RTCP Bandwidth Scaling

 RTCP attempts to limit its traffic to 5% of session bandwidth.

Example

 Suppose one sender,

sending video at 2 Mbps.

Then RTCP attempts to limit its traffic to 100 Kbps.

 RTCP gives 75% of rate to receivers; remaining 25%

to sender

 75 kbps is equally shared among receivers:

 with R receivers, each

receiver gets to send RTCP traffic at 75/R kbps.

 sender gets to send RTCP traffic at 25 kbps.

 participant determines RTCP packet transmission period by calculating avg RTCP packet size (across entire session) and dividing by allocated rate

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SIP: Session Initiation Protocol

[RFC 3261]

SIP long-term vision:

 all telephone calls, video conference calls take place over Internet

 people are identified by names or e-mail addresses, rather than by phone numbers

 you can reach callee, no matter where callee

roams, no matter what IP device callee is currently using

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SIP Services

 Setting up a call, SIP provides mechanisms ..

 for caller to let callee know she

wants to establish a call

 so caller, callee can agree on media type, encoding

 to end call

 determine current IP address of callee:

 maps mnemonic

identifier to current IP address

 call management:

 add new media streams during call

 change encoding during call

 invite others

 transfer, hold calls

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Setting up a call to known IP address

 Alice’s SIP invite message indicates her port number, IP address, encoding she prefers to receive (PCM ulaw)

 Bob’s 200 OK message indicates his port number, IP address, preferred

encoding (GSM)

 SIP messages can be sent over TCP or UDP;

here sent over RTP/UDP.

default SIP port number is 5060.

time time

Bob's

terminal rings Alice

167.180.112.24

Bob

193.64.210.89

port 5060

port 38060

m Law audio

GSM

port 48753 INVITE bob@193.64.210.89

c=IN IP4 167.180.112.24 m=audio 38060 RTP/AVP 0

port 5060 200 OK

c=IN IP4 193.64.210.89 m=audio 48753 RTP/AVP 3

ACK port 5060

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Setting up a call (more)

 codec negotiation:

 suppose Bob doesn’t have PCM ulaw

encoder.

 Bob will instead reply with 606 Not

Acceptable Reply, listing his encoders Alice can then send new INVITE

message, advertising different encoder

 rejecting a call

 Bob can reject with replies “busy,”

“gone,” “payment required,”

“forbidden”

 media can be sent over RTP or some other

protocol

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Example of SIP message

INVITE sip:bob@domain.com SIP/2.0 Via: SIP/2.0/UDP 167.180.112.24 From: sip:alice@hereway.com

To: sip:bob@domain.com

Call-ID: a2e3a@pigeon.hereway.com Content-Type: application/sdp

Content-Length: 885

c=IN IP4 167.180.112.24 m=audio 38060 RTP/AVP 0 Notes:

 HTTP message syntax

 sdp = session description protocol

 Call-ID is unique for every call.

 Here we don’t know Bob’s IP address.

Intermediate SIP servers needed.

 Alice sends, receives SIP messages using SIP default port 506

 Alice specifies in Via:

header that SIP client sends, receives SIP

messages over UDP

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Name translation and user locataion

 caller wants to call callee, but only has

callee’s name or e-mail address.

 need to get IP address of callee’s current

host:

 user moves around

 DHCP protocol

 user has different IP devices (PC, PDA, car device)

 result can be based on:

 time of day (work, home)

 caller (don’t want boss to call you at home)

 status of callee (calls sent to voicemail when callee is already talking to

someone)

Service provided by SIP servers:

 SIP registrar server

 SIP proxy server

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SIP Registrar

REGISTER sip:domain.com SIP/2.0 Via: SIP/2.0/UDP 193.64.210.89 From: sip:bob@domain.com

To: sip:bob@domain.com Expires: 3600

 when Bob starts SIP client, client sends SIP REGISTER message to Bob’s registrar server (similar function needed by Instant Messaging) Register Message:

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SIP Proxy

 Alice sends invite message to her proxy server

 contains address sip:bob@domain.com

 proxy responsible for routing SIP messages to callee

 possibly through multiple proxies.

 callee sends response back through the same set of proxies.

 proxy returns SIP response message to Alice

 contains Bob’s IP address

 proxy analogous to local DNS server

(69)

Example

Caller jim@umass.edu with places a

call to keith@upenn.edu (1) Jim sends INVITE message to umass SIP

proxy. (2) Proxy forwards request to upenn

registrar server.

(3) upenn server returns redirect response,

indicating that it should try keith@eurecom.fr

(4) umass proxy sends INVITE to eurecom registrar. (5) eurecom

registrar forwards INVITE to 197.87.54.21, which is running keith’s SIP client. (6-8) SIP response sent back (9) media sent directly

between clients.

Note: also a SIP ack message, which is not shown.

SIP client 217.123.56.89

SIP client 197.87.54.21 SIP proxy

umass.edu

SIP registrar upenn.edu

SIP registrar eurecom.fr

1

2

3 4

5

6 7

8

9

(70)

Comparison with H.323

 H.323 is another signaling protocol for real-time, interactive

 H.323 is a complete,

vertically integrated suite of protocols for multimedia conferencing: signaling,

registration, admission control, transport, codecs

 SIP is a single component.

Works with RTP, but does not mandate it. Can be combined with other protocols, services

 H.323 comes from the ITU (telephony).

 SIP comes from IETF:

Borrows much of its concepts from HTTP

 SIP has Web flavor, whereas H.323 has telephony flavor.

 SIP uses the KISS

principle: Keep it simple stupid.

(71)

Chapter 7 outline

RTP, RTCP, SIP

guarantees

(72)

Providing Multiple Classes of Service

 thus far: making the best of best effort service

 one-size fits all service model

 alternative: multiple classes of service

 partition traffic into classes

 network treats different classes of traffic

differently (analogy: VIP service vs regular service)

0111

 granularity:

differential service among multiple

classes, not among individual

connections

 history: ToS bits

(73)

Multiple classes of service: scenario

R1 R2

H1

H2

H3

1.5 Mbps link H4 R1 output

interface queue

(74)

Scenario 1: mixed FTP and audio

 Example: 1Mbps IP phone, FTP share 1.5 Mbps link.

 bursts of FTP can congest router, cause audio loss

 want to give priority to audio over FTP

packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly

Principle 1

R1 R2

(75)

Principles for QOS Guarantees (more)

 what if applications misbehave (audio sends higher than declared rate)

 policing: force source adherence to bandwidth allocations

 marking and policing at network edge:

 similar to ATM UNI (User Network Interface)

provide protection (isolation) for one class from others Principle 2

R1 R2

1.5 Mbps link 1 Mbps

phone

packet marking and policing

(76)

Principles for QOS Guarantees (more)

 Allocating fixed (non-sharable) bandwidth to flow:

inefficient use of bandwidth if flows doesn’t use its allocation

While providing isolation, it is desirable to use resources as efficiently as possible

Principle 3

R1 R2

phone

1 Mbps logical link

0.5 Mbps logical link

(77)

Scheduling And Policing Mechanisms

 scheduling: choose next packet to send on link

 FIFO (first in first out) scheduling: send in order of arrival to queue

 real-world example?

 discard policy: if packet arrives to full queue: who to discard?

• Tail drop: drop arriving packet

• priority: drop/remove on priority basis

• random: drop/remove randomly

(78)

Scheduling Policies: more

Priority scheduling: transmit highest priority queued packet

 multiple classes, with different priorities

 class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc..

 Real world example?

(79)

Scheduling Policies: still more

round robin scheduling:

 multiple classes

 cyclically scan class queues, serving one from each class (if available)

 real world example?

(80)

Scheduling Policies: still more

Weighted Fair Queuing:

 generalized Round Robin

 each class gets weighted amount of service in each cycle

 real-world example?

(81)

Policing Mechanisms

Goal:

limit traffic to not exceed declared parameters Three common-used criteria:

 (Long term) Average Rate: how many pkts can be sent per unit time (in the long run)

 crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average!

 Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 1500 ppm peak rate

 (Max.) Burst Size: max. number of pkts sent consecutively (with no intervening idle)

(82)

Policing Mechanisms

Token Bucket:

limit input to specified Burst Size and Average Rate.

 bucket can hold b tokens

 tokens generated at rate r token/sec unless bucket full

 over interval of length t: number of packets admitted less than or equal to (r t + b).

(83)

Policing Mechanisms (more)

 token bucket, WFQ combine to provide guaranteed upper bound on delay, i.e., QoS guarantee!

WFQ

token rate, r bucket size, b

per-flow rate, R D = b/R max

arriving traffic

(84)

IETF Differentiated Services

 want “qualitative” service classes

 “behaves like a wire”

 relative service distinction: Platinum, Gold, Silver

 scalability: simple functions in network core,

relatively complex functions at edge routers (or hosts)

 signaling, maintaining per-flow router state difficult with large number of flows

 don’t define define service classes, provide functional components to build service classes

(85)

Edge router:

 per-flow traffic management

 marks packets as in-profile and out-profile

Core router:

 per class traffic management

 buffering and scheduling based on marking at edge

 preference given to in-profile packets

Diffserv Architecture

scheduling

. . .

r b

marking

(86)

Edge-router Packet Marking

 class-based marking: packets of different classes marked differently

 intra-class marking: conforming portion of flow marked differently than non-conforming one

 profile: pre-negotiated rate A, bucket size B

 packet marking at edge based on per-flow profile

Possible usage of marking:

User packets

Rate A B

(87)

Classification and Conditioning

 Packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6

 6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that the packet will receive

 2 bits are currently unused

(88)

Classification and Conditioning

may be desirable to limit traffic injection rate of some class:

 user declares traffic profile (e.g., rate, burst size)

 traffic metered, shaped if non-conforming

(89)

Forwarding (PHB)

 PHB result in a different observable (measurable) forwarding performance behavior

 PHB does not specify what mechanisms to use to ensure required PHB performance behavior

 Examples:

 Class A gets x% of outgoing link bandwidth over time intervals of a specified length

 Class A packets leave first before packets from class B

(90)

Forwarding (PHB)

PHBs being developed:

 Expedited Forwarding: pkt departure rate of a class equals or exceeds specified rate

 logical link with a minimum guaranteed rate

 Assured Forwarding: 4 classes of traffic

 each guaranteed minimum amount of bandwidth

 each with three drop preference partitions

(91)

Chapter 7 outline

RTP, RTCP, SIP

guarantees

(92)

Chapter 7 outline

 7.1 Multimedia

Networking Applications

 7.2 Streaming stored audio and video

 7.3 Real-time Multimedia:

Internet Phone study

 7.4 Protocols for Real- Time Interactive

Applications

 RTP,RTCP,SIP

 7.5 Distributing

Multimedia: content distribution networks

 7.6 Beyond Best Effort

 7.7 Scheduling and Policing Mechanisms

 7.8 Integrated Services and Differentiated Services

 7.9 RSVP

(93)

Principles for QOS Guarantees (more)

 Basic fact of life: can not support traffic demands beyond link capacity

Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs

Principle 4

R1 R2

phone

1 Mbps phone

(94)

QoS guarantee scenario

 Resource reservation

 call setup, signaling (RSVP)

 traffic, QoS declaration

 per-element admission control

 QoS-sensitive scheduling (e.g.,

WFQ)

request/

reply

(95)

IETF Integrated Services

 architecture for providing QOS guarantees in IP networks for individual application sessions

 resource reservation: routers maintain state info (a la VC) of allocated resources, QoS req’s

 admit/deny new call setup requests:

Question: can newly arriving flow be admitted with performance guarantees while not violated QoS guarantees made to already admitted flows?

(96)

Call Admission

Arriving session must :

 declare its QOS requirement

 R-spec: defines the QOS being requested

 characterize traffic it will send into network

 T-spec: defines traffic characteristics

 signaling protocol: needed to carry R-spec and T- spec to routers (where reservation is required)

 RSVP

(97)

Intserv QoS: Service models

[rfc2211, rfc 2212]

Guaranteed service:

 worst case traffic arrival:

leaky-bucket-policed source

 simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988]

Controlled load service:

 "a quality of service closely approximating the QoS that same flow would receive

from an unloaded network element."

WFQ

token rate, r bucket size, b

per-flow rate, R D = b/R max

arriving traffic